MediaPortal Audio renderer - better video playback quality (7 Viewers)

[Airshark]

AW: MediaPortal Audio renderer - better video playback quality

Does this renderer only run in Windows7 and EVR?
Cause I have WinXP and no EVR... so when I push SHIFT-1 I only get a smal statistic, and not graphical?

 

[arion_p]

All looks good, but why is there a '2*' in the last equation - it's presumably meant to be the average of all channels minus the target (j) channel ? (is it to make sure it becomes the 'dominant' channel ?)

Q(t) = Sum(Ci(t), i=1..N)/N

SoundTouch uses Q(t) = (L(t)+R(t))/2

We need to solve the above system for R(t) (since we assign L(t) = Cj(t)) and the solution is:

Q(t) = Sum(Ci(t), i=1..N)/N
Q(t) = (L(t)+R(t))/2
<=>
Q(t) = Sum(Ci(t), i=1..N)/N
Q(t) = (Cj(t)+R(t))/2
<=>
Sum(Ci(t), i=1..N)/N = (Cj(t)+R(t))/2
<=>
2*Sum(Ci(t), i=1..N)/N = (Cj(t)+R(t))
<=>
2*Sum(Ci(t), i=1..N)/N - Cj(t) = R(t)
<=>
R(t) = 2*Sum(Ci(t), i=1..N)/N - Cj(t)

 

[tourettes]

First of all, currently we are only time-stretching, not resampling the audio. I do not know if we will need to resample, but so far I see no need.

Imho the time-stretching is better than resampling. I prefer a pitch coherent sound even with worst quality.

Actually I wouldn't ever consider using resampling to try to change the lenght of audio stream, the results would be just horrible

Instead there are few other uses for it:

1) upsampling the source. 48 to 96Khz for example. Althou I consider it just useless creation of random "noise" on high frequencies probably just coloring the audio to sound unnaturally bright (of course there are people who thing that would sound alreays better than the original )

2) To overcome some broken HW resampling (coug... Creative... cough) Some HW has just really bad resampling and for example 44.1Khz could be resampled with the audio renderer to the 48Khz that might be only "real" unmolested frequency on some HW.

 

[tourettes]

Re: AW: MediaPortal Audio renderer - better video playback quality

Does this renderer only run in Windows7 and EVR?
Cause I have WinXP and no EVR... so when I push SHIFT-1 I only get a smal statistic, and not graphical?

Audio renderer itself works with the Windows XP, but the video renderer and audio renderer interoperatibility will require EVR (since no code has been added for VMR9, and probably wont be added). So, basicly there is no point in Windows XP testing, at least currently.

 

[arion_p]

1) upsampling the source. 48 to 96Khz for example. Althou I consider it just useless creation of random "noise" on high frequencies probably just coloring the audio to sound unnaturally bright (of course there are people who thing that would sound alreays better than the original )

Normally descent D/A converters will do at least 3rd order oversampling to at least 96Khz or even 192Khz (while Sigma-Delta converters usually oversample to at least 768Khz). If however the D/A does not do any oversampling, or does it poorly, doing our own upsampling might help a bit. To understand why it sounds better, consider a sine wave at 16Khz sampled at 48Khz, a single period of the sine wave consists of only 3 samples! The end result is a square wave with severe intermodulation distortion. Now if you resample that to 192Kh using a high quality 3rd or 4th order filter or even better using wavelets the result is 12 samples per period that more closely match the sine wave.

That being said, a high quality resampler is hard to implement, especially in real-time. So I still think resampling in the audio renderer is not much use and it is better left to be done by a real DSP in the amp.

 

[tourettes]

1) upsampling the source. 48 to 96Khz for example. Althou I consider it just useless creation of random "noise" on high frequencies probably just coloring the audio to sound unnaturally bright (of course there are people who thing that would sound alreays better than the original )

Normally descent D/A converters will do at least 3rd order oversampling to at least 96Khz or even 192Khz (while Sigma-Delta converters usually oversample to at least 768Khz). If however the D/A does not do any oversampling, or does it poorly, doing our own upsampling might help a bit. To understand why it sounds better, consider a sine wave at 16Khz sampled at 48Khz, a single period of the sine wave consists of only 3 samples! The end result is a square wave with severe intermodulation distortion. Now if you resample that to 192Kh using a high quality 3rd or 4th order filter or even better using wavelets the result is 12 samples per period that more closely match the sine wave.

For a sine wave indeed resampling would give much more detail since it is quite easy to "forecast" where the missing details should go, but for something like distorted guitar sound you could be creating quite easy some false data that would be further away form the real recorded sound than the one the less detailed "real" sampled sound is so it could end up sounding worser (just like resampling video can create rigging effect to the video where edges start to glow).

 

[grubi]

AW: MediaPortal Audio renderer - better video playback quality

Hopefully bitstream support will be implemented sometime in the future although it is marked low priority (would be glad to join for testing it).
What is the reason for that?
Is it difficult to implement or considered useless (it's clear that the speedup/slowdown feature will not make sense here)?
IMHO bitstreaming sound is quite common and always a good idea if you are after high quality sound.

Cheers
grubi.

 

[tourettes]

Hopefully bitstream support will be implemented sometime in the future although it is marked low priority (would be glad to join for testing it).
What is the reason for that?
Is it difficult to implement or considered useless (it's clear that the speedup/slowdown feature will not make sense here)?
IMHO bitstreaming sound is quite common and always a good idea if you are after high quality sound.

Cheers
grubi.

bitsreaming indeed would disable all the video playback related quality improvements. Also the difference between bitstreamed audio and lpcm is non-existing since we are talking about lossless audio formats and decoding those will always produce the 1:1 result. So implementing the bitstreaming support has the lowest priority.
Posted via Mobile Device

 

[grubi]

AW: MediaPortal Audio renderer - better video playback quality

bitsreaming indeed would disable all the video playback related quality improvements

Really all?
What about eliminating the clock drift problem?

 

[jameson_uk]

Also the difference between bitstreamed audio and lpcm is non-existing since we are talking about lossless audio formats and decoding those will always produce the 1:1 result. So implementing the bitstreaming support has the lowest priority.

With a couple of issues
(a) there is no directshow decoder for DTS-HD so only option for that is bitstreaming or downgrading it to the core standard DTS audio
(b) some directshow decoders for TrueHD actually resample everything to 16 bit / 48Khz when converting to LPCM. I have a feeling they did sort this out in recent SVN versions of FFDShow but still the case in commercial players like TMT and PowerDVD (and therefore in Directshow if you use their filters)